1. Field of the Invention
The present general inventive concept relates to an apparatus and method of encoding and decoding an input signal, and more particularly, to a method and apparatus to quantize and dequantize an input signal to obtain better compression efficiency, and a method and apparatus to encode and decode an input signal.
2. Description of the Related Art
An input signal containing information is originally an analog signal having a continuous waveform in terms of amplitude and time. Thus, in order for the waveform to be expressed as a discrete signal, analog-to-digital (A/D) conversion is required. The A/D conversion includes two processes. First, a sampling process is used to convert a time-continuous signal into a discrete signal, and secondly, an amplitude quantizing process is used to limit the number of feasible amplitudes to a finite number. In the amplitude quantizing process, an input amplitude x(n) is converted into an element y(n) included in a finite set of feasible amplitudes at time n.
With the recent development of digital signal processing techniques, a digital audio signal storing/restoring method is widely used by ordinary users, in which a conventional analog signal is sampled and quantized, converted into pulse code modulation (PCM) data, that is, a digital signal, and is stored in a recording/storing medium such as a compact disc (CD) or a digital audio tape (DAT), so that a user can reproduce the stored signal if necessary. According to the digital audio storing/restoring method, audio quality has been improved, and deterioration of audio quality due to storage time has been overcome in comparison with an analog type method such as a long-play record (LP) or tape. However, due to a large size of digital data, there has been a problem in signal storing and transfer processes.
In order to solve this problem, a differential pulse code modulation (DPCM) method, an adaptive differential pulse code modulation (ADPCM) method, and so on, have been developed in order to compress a digital audio signal. By using these methods, although an effort has been made to obtain a digital audio signal having a small data size, there is a problem in that efficiency greatly depends on signal types. A Moving Pictures Expert Group (MPEG)/audio scheme recently standardized by the International Standard Organization (ISO) or an Audio Codec (AC-2/AC-3) scheme developed by Dolby use psychoacoustic modeling to reduce a data size. These methods have significantly reduced the data size in an effective manner regardless of signal characteristics.
In the conventional audio signal compression technique, such as MPEG-1/audio, MPEG-2/audio, or AC-2/AC-3, a time-domain signal is bound to a block having a specific size, and is transformed into a frequency-domain signal. Thereafter, the transformed signal is scalar-quantized by using the psychoacoustic modeling. The quantization scheme is simple, but is not an optimum scheme, even if an input sample is statistically independent. If the input sample is statistically dependent, the quantization scheme is far less sufficient. To avoid this problem, lossless-encoding, such as entropy encoding, is performed. Alternatively, encoding is performed along with a certain type of adaptive quantization. Accordingly, an encoding process has become significantly complex in comparison with a method in which only simple PCM data is stored. Furthermore, a bit stream includes not only quantized PCM data but also additional information for compressing a signal.
The MPEG/audio standard or the AC-2/AC-3 technique provides audio quality which is almost the same as that of a compact disc (CD) at a bit rate ranging from 64 Kbps to 384 Kbps, which is ⅙ to ⅛ smaller than that of conventional digital encoding. Thus, the MPEG/audio standard is expected to play an important role in storing and transferring an audio signal of various systems, such as digital audio broadcasting (DAB), internet phones, audio on demand (AOD), and multimedia systems.
In an MPEG-1/2 audio encoding technique, sub-band filtering is performed, and a sub-band sample is then linear-quantized by using bit assignment information proposed by the psychoacoustic modeling. Thereafter, the sub-band sample is subjected to a bit packing process, thereby completing encoding. In the quantizing process, a linear quantizer has optimum capability when a data distribution is uniform. However, the linear quantizer causes severe signal distortion when a number of assigned bits is small.
An actual data distribution is not uniform but is similar to a Gaussian or Laplacian distribution. In this case, a quantizer may be designed to fit a corresponding distribution, so as to obtain an optimum result in terms of mean squared error (MSE). A general audio encoder such as advanced audio coding (AAC) of MPEG-2/4 uses a nonlinear quantizer. This is designed by taking a sample distribution of a modified discrete cosine transform (MDCT) and a psychoacoustic aspect into account. However, since one quantizer is used for a variety of input signals, there are disadvantages in that encoding cannot be effectively achieved in terms of a bit rate, and audio quality may deteriorate. Furthermore, there is an increase in complexity in order to obtain high audio quality.